Changeset 212401 in webkit
- Timestamp:
- Feb 15, 2017 3:23:30 PM (7 years ago)
- Location:
- trunk/Source/ThirdParty/libwebrtc
- Files:
-
- 1 added
- 29 edited
Legend:
- Unmodified
- Added
- Removed
-
trunk/Source/ThirdParty/libwebrtc/ChangeLog
r212397 r212401 1 2017-02-15 Alex Christensen <achristensen@webkit.org> 2 3 Make libwebrtc.dylib 4 https://bugs.webkit.org/show_bug.cgi?id=168335 5 6 Reviewed by Dan Bernstein. 7 8 We were building libwebrtc as a static library, which would prevent us from weak linking with it. 9 We need to explicitly export what we use from WebCore or WebKit2, and RTCLogging.mm now needs to 10 be built on Mac, so we make it not automatically reference counted to make it work on 32-bit El Capitan. 11 12 * Configurations/libwebrtc.xcconfig: 13 * Source/webrtc/api/jsep.h: 14 * Source/webrtc/api/mediastream.h: 15 * Source/webrtc/api/notifier.h: 16 (webrtc::Notifier::Notifier): Deleted. 17 (webrtc::Notifier::RegisterObserver): Deleted. 18 (webrtc::Notifier::UnregisterObserver): Deleted. 19 (webrtc::Notifier::FireOnChanged): Deleted. 20 * Source/webrtc/api/peerconnectioninterface.h: 21 * Source/webrtc/base/asyncpacketsocket.h: 22 * Source/webrtc/base/asyncresolverinterface.h: 23 (rtc::AsyncResolverInterface::address): Deleted. 24 * Source/webrtc/base/copyonwritebuffer.h: 25 (rtc::CopyOnWriteBuffer::CopyOnWriteBuffer): Deleted. 26 (rtc::CopyOnWriteBuffer::data): Deleted. 27 (rtc::CopyOnWriteBuffer::cdata): Deleted. 28 (rtc::CopyOnWriteBuffer::size): Deleted. 29 (rtc::CopyOnWriteBuffer::capacity): Deleted. 30 (rtc::CopyOnWriteBuffer::operator=): Deleted. 31 (rtc::CopyOnWriteBuffer::operator!=): Deleted. 32 (rtc::CopyOnWriteBuffer::operator[]): Deleted. 33 (rtc::CopyOnWriteBuffer::SetData): Deleted. 34 (rtc::CopyOnWriteBuffer::AppendData): Deleted. 35 (rtc::CopyOnWriteBuffer::swap): Deleted. 36 (rtc::CopyOnWriteBuffer::IsConsistent): Deleted. 37 * Source/webrtc/base/event.h: 38 * Source/webrtc/base/export.h: Added. 39 * Source/webrtc/base/helpers.h: 40 * Source/webrtc/base/ipaddress.h: 41 (rtc::IPAddress::IPAddress): Deleted. 42 (rtc::IPAddress::~IPAddress): Deleted. 43 (rtc::IPAddress::operator=): Deleted. 44 (rtc::IPAddress::family): Deleted. 45 * Source/webrtc/base/location.h: 46 (rtc::Location::function_name): Deleted. 47 (rtc::Location::file_and_line): Deleted. 48 * Source/webrtc/base/messagehandler.h: 49 (rtc::MessageHandler::MessageHandler): Deleted. 50 * Source/webrtc/base/network.h: 51 (rtc::NetworkManagerBase::ipv6_enabled): Deleted. 52 (rtc::NetworkManagerBase::set_ipv6_enabled): Deleted. 53 (rtc::NetworkManagerBase::set_max_ipv6_networks): Deleted. 54 (rtc::NetworkManagerBase::max_ipv6_networks): Deleted. 55 (rtc::NetworkManagerBase::set_enumeration_permission): Deleted. 56 (rtc::BasicNetworkManager::started): Deleted. 57 (rtc::BasicNetworkManager::set_network_ignore_list): Deleted. 58 (rtc::BasicNetworkManager::set_ignore_non_default_routes): Deleted. 59 (rtc::Network::default_local_address_provider): Deleted. 60 (rtc::Network::set_default_local_address_provider): Deleted. 61 (rtc::Network::name): Deleted. 62 (rtc::Network::description): Deleted. 63 (rtc::Network::prefix): Deleted. 64 (rtc::Network::prefix_length): Deleted. 65 (rtc::Network::key): Deleted. 66 (rtc::Network::ip): Deleted. 67 (rtc::Network::AddIP): Deleted. 68 (rtc::Network::GetIPs): Deleted. 69 (rtc::Network::ClearIPs): Deleted. 70 (rtc::Network::scope_id): Deleted. 71 (rtc::Network::set_scope_id): Deleted. 72 (rtc::Network::ignored): Deleted. 73 (rtc::Network::set_ignored): Deleted. 74 (rtc::Network::type): Deleted. 75 (rtc::Network::set_type): Deleted. 76 (rtc::Network::GetCost): Deleted. 77 (rtc::Network::id): Deleted. 78 (rtc::Network::set_id): Deleted. 79 (rtc::Network::preference): Deleted. 80 (rtc::Network::set_preference): Deleted. 81 (rtc::Network::active): Deleted. 82 (rtc::Network::set_active): Deleted. 83 * Source/webrtc/base/proxyinfo.h: 84 * Source/webrtc/base/refcountedobject.h: 85 (rtc::RefCountedObject::RefCountedObject): Deleted. 86 (rtc::RefCountedObject::AddRef): Deleted. 87 (rtc::RefCountedObject::Release): Deleted. 88 (rtc::RefCountedObject::HasOneRef): Deleted. 89 (rtc::RefCountedObject::~RefCountedObject): Deleted. 90 * Source/webrtc/base/socketaddress.h: 91 (rtc::SocketAddress::hostname): Deleted. 92 (rtc::SocketAddress::family): Deleted. 93 (rtc::SocketAddress::scope_id): Deleted. 94 (rtc::SocketAddress::SetScopeID): Deleted. 95 (rtc::SocketAddress::operator !=): Deleted. 96 * Source/webrtc/base/thread.h: 97 * Source/webrtc/common_types.h: 98 * Source/webrtc/common_video/include/video_frame_buffer.h: 99 (webrtc::I420Buffer::Copy): Deleted. 100 (webrtc::I420Buffer::CropAndScaleFrom): Deleted. 101 (webrtc::I420Buffer::ScaleFrom): Deleted. 102 * Source/webrtc/common_video/libyuv/include/webrtc_libyuv.h: 103 * Source/webrtc/p2p/base/basicpacketsocketfactory.h: 104 * Source/webrtc/p2p/client/basicportallocator.h: 105 (cricket::BasicPortAllocator::network_ignore_mask): Deleted. 106 (cricket::BasicPortAllocator::network_manager): Deleted. 107 (cricket::BasicPortAllocator::socket_factory): Deleted. 108 * Source/webrtc/sdk/objc/Framework/Classes/RTCLogging.mm: 109 (RTCFileName): 110 * Source/webrtc/sdk/objc/Framework/Classes/videotoolboxvideocodecfactory.h: 111 * Source/webrtc/video_frame.h: 112 (webrtc::VideoFrame::timestamp_us): Deleted. 113 (webrtc::VideoFrame::set_timestamp_us): Deleted. 114 (webrtc::VideoFrame::set_timestamp): Deleted. 115 (webrtc::VideoFrame::timestamp): Deleted. 116 (webrtc::VideoFrame::transport_frame_id): Deleted. 117 (webrtc::VideoFrame::set_ntp_time_ms): Deleted. 118 (webrtc::VideoFrame::ntp_time_ms): Deleted. 119 (webrtc::VideoFrame::rotation): Deleted. 120 (webrtc::VideoFrame::set_rotation): Deleted. 121 (webrtc::VideoFrame::set_render_time_ms): Deleted. 122 (webrtc::VideoFrame::render_time_ms): Deleted. 123 (webrtc::VideoFrame::is_texture): Deleted. 124 * build: Added. 125 * build/Debug: Added. 126 * libwebrtc.xcodeproj/project.pbxproj: 127 1 128 2017-02-15 Youenn Fablet <youenn@apple.com> 2 129 -
trunk/Source/ThirdParty/libwebrtc/Configurations/libwebrtc.xcconfig
r212397 r212401 1 PRODUCT_NAME = webrtc;1 PRODUCT_NAME = libwebrtc; 2 2 3 3 CLANG_WARN_BOOL_CONVERSION = YES; … … 22 22 EXCLUDED_SOURCE_FILE_NAMES[sdk=iphoneos*] = *_sse.cc *_sse2.cc macutils.cc macwindowpicker.cc audio_device_mac.cc audio_mixer_manager_mac.cc; 23 23 EXCLUDED_SOURCE_FILE_NAMES[sdk=iphonesimulator*] = macutils.cc macwindowpicker.cc audio_device_mac.cc audio_mixer_manager_mac.cc; 24 EXCLUDED_SOURCE_FILE_NAMES[sdk=macosx*] = audio_device_ios.mm voice_processing_audio_unit.mm audio_device_not_implemented_ios.mm RTCAudioSessionConfiguration.m RTCAudioSessionDelegateAdapter.mm RTCAudioSession.mm RTCAudioSession+Configuration.mm RTCUIApplication.mm RTCLogging.mm;24 EXCLUDED_SOURCE_FILE_NAMES[sdk=macosx*] = audio_device_ios.mm voice_processing_audio_unit.mm audio_device_not_implemented_ios.mm RTCAudioSessionConfiguration.m RTCAudioSessionDelegateAdapter.mm RTCAudioSession.mm RTCAudioSession+Configuration.mm RTCUIApplication.mm; -
trunk/Source/ThirdParty/libwebrtc/Source/webrtc/api/jsep.h
r212333 r212401 18 18 19 19 #include "webrtc/base/basictypes.h" 20 #include "webrtc/base/export.h" 20 21 #include "webrtc/base/refcount.h" 21 22 … … 56 57 // Returns NULL if the sdp string can't be parsed. 57 58 // |error| can be NULL if doesn't care about the failure reason. 58 IceCandidateInterface* CreateIceCandidate(const std::string& sdp_mid,59 WEBRTC_EXPORT IceCandidateInterface* CreateIceCandidate(const std::string& sdp_mid, 59 60 int sdp_mline_index, 60 61 const std::string& sdp, … … 113 114 // Returns NULL if the sdp string can't be parsed or the type is unsupported. 114 115 // |error| can be NULL if doesn't care about the failure reason. 115 SessionDescriptionInterface* CreateSessionDescription(const std::string& type,116 WEBRTC_EXPORT SessionDescriptionInterface* CreateSessionDescription(const std::string& type, 116 117 const std::string& sdp, 117 118 SdpParseError* error); -
trunk/Source/ThirdParty/libwebrtc/Source/webrtc/api/mediastream.h
r212333 r212401 19 19 #include "webrtc/api/mediastreaminterface.h" 20 20 #include "webrtc/api/notifier.h" 21 #include "webrtc/base/export.h" 21 22 22 23 namespace webrtc { 23 24 24 class MediaStream : public Notifier<MediaStreamInterface> {25 class WEBRTC_EXPORT MediaStream : public Notifier<MediaStreamInterface> { 25 26 public: 26 27 static rtc::scoped_refptr<MediaStream> Create(const std::string& label); -
trunk/Source/ThirdParty/libwebrtc/Source/webrtc/api/notifier.h
r212333 r212401 16 16 #include "webrtc/api/mediastreaminterface.h" 17 17 #include "webrtc/base/common.h" 18 #include "webrtc/base/export.h" 18 19 19 20 namespace webrtc { … … 21 22 // Implement a template version of a notifier. 22 23 template <class T> 23 class Notifier : public T {24 class WEBRTC_EXPORT Notifier : public T { 24 25 public: 25 26 Notifier() { -
trunk/Source/ThirdParty/libwebrtc/Source/webrtc/api/peerconnectioninterface.h
r212333 r212401 66 66 #include "webrtc/api/statstypes.h" 67 67 #include "webrtc/api/umametrics.h" 68 #include "webrtc/base/export.h" 68 69 #include "webrtc/base/fileutils.h" 69 70 #include "webrtc/base/network.h" … … 740 741 // If non-null, ownership of |default_adm|, |encoder_factory| and 741 742 // |decoder_factory| are transferred to the returned factory. 742 rtc::scoped_refptr<PeerConnectionFactoryInterface> CreatePeerConnectionFactory(743 WEBRTC_EXPORT rtc::scoped_refptr<PeerConnectionFactoryInterface> CreatePeerConnectionFactory( 743 744 rtc::Thread* network_thread, 744 745 rtc::Thread* worker_thread, -
trunk/Source/ThirdParty/libwebrtc/Source/webrtc/base/asyncpacketsocket.h
r212333 r212401 14 14 #include "webrtc/base/constructormagic.h" 15 15 #include "webrtc/base/dscp.h" 16 #include "webrtc/base/export.h" 16 17 #include "webrtc/base/sigslot.h" 17 18 #include "webrtc/base/socket.h" … … 23 24 // extension, including the information needed to update the authentication tag 24 25 // after changing the value. 25 struct PacketTimeUpdateParams {26 struct WEBRTC_EXPORT PacketTimeUpdateParams { 26 27 PacketTimeUpdateParams(); 27 28 ~PacketTimeUpdateParams(); … … 66 67 // Provides the ability to receive packets asynchronously. Sends are not 67 68 // buffered since it is acceptable to drop packets under high load. 68 class AsyncPacketSocket : public sigslot::has_slots<> {69 class WEBRTC_EXPORT AsyncPacketSocket : public sigslot::has_slots<> { 69 70 public: 70 71 enum State { -
trunk/Source/ThirdParty/libwebrtc/Source/webrtc/base/asyncresolverinterface.h
r212333 r212401 12 12 #define WEBRTC_BASE_ASYNCRESOLVERINTERFACE_H_ 13 13 14 #include "webrtc/base/export.h" 14 15 #include "webrtc/base/sigslot.h" 15 16 #include "webrtc/base/socketaddress.h" … … 18 19 19 20 // This interface defines the methods to resolve the address asynchronously. 20 class AsyncResolverInterface {21 class WEBRTC_EXPORT AsyncResolverInterface { 21 22 public: 22 23 AsyncResolverInterface(); -
trunk/Source/ThirdParty/libwebrtc/Source/webrtc/base/checks.h
r210942 r212401 12 12 #define WEBRTC_BASE_CHECKS_H_ 13 13 14 #include "webrtc/base/export.h" 14 15 #include "webrtc/typedefs.h" 15 16 … … 211 212 212 213 // This is identical to LogMessageVoidify but in name. 213 class FatalMessageVoidify {214 class WEBRTC_EXPORT FatalMessageVoidify { 214 215 public: 215 216 FatalMessageVoidify() { } -
trunk/Source/ThirdParty/libwebrtc/Source/webrtc/base/copyonwritebuffer.h
r212333 r212401 17 17 #include "webrtc/base/buffer.h" 18 18 #include "webrtc/base/checks.h" 19 #include "webrtc/base/export.h" 19 20 #include "webrtc/base/refcount.h" 20 21 #include "webrtc/base/scoped_ref_ptr.h" … … 22 23 namespace rtc { 23 24 24 class CopyOnWriteBuffer {25 class WEBRTC_EXPORT CopyOnWriteBuffer { 25 26 public: 26 27 // An empty buffer. -
trunk/Source/ThirdParty/libwebrtc/Source/webrtc/base/event.h
r212333 r212401 21 21 22 22 #include "webrtc/base/basictypes.h" 23 #include "webrtc/base/export.h" 23 24 24 25 namespace rtc { 25 26 26 class Event {27 class WEBRTC_EXPORT Event { 27 28 public: 28 29 static const int kForever = -1; -
trunk/Source/ThirdParty/libwebrtc/Source/webrtc/base/helpers.h
r212333 r212401 14 14 #include <string> 15 15 #include "webrtc/base/basictypes.h" 16 #include "webrtc/base/export.h" 16 17 17 18 namespace rtc { … … 26 27 // Generates a (cryptographically) random string of the given length. 27 28 // We generate base64 values so that they will be printable. 28 std::string CreateRandomString(size_t length);29 WEBRTC_EXPORT std::string CreateRandomString(size_t length); 29 30 30 31 // Generates a (cryptographically) random string of the given length. -
trunk/Source/ThirdParty/libwebrtc/Source/webrtc/base/ipaddress.h
r212333 r212401 28 28 #include "webrtc/base/basictypes.h" 29 29 #include "webrtc/base/byteorder.h" 30 #include "webrtc/base/export.h" 30 31 #if defined(WEBRTC_WIN) 31 32 #include "webrtc/base/win32.h" … … 48 49 49 50 // Version-agnostic IP address class, wraps a union of in_addr and in6_addr. 50 class IPAddress {51 class WEBRTC_EXPORT IPAddress { 51 52 public: 52 53 IPAddress() : family_(AF_UNSPEC) { -
trunk/Source/ThirdParty/libwebrtc/Source/webrtc/base/location.h
r212333 r212401 14 14 #include <string> 15 15 16 #include "webrtc/base/export.h" 16 17 #include "webrtc/system_wrappers/include/stringize_macros.h" 17 18 … … 22 23 // This is a stripped down version of: 23 24 // https://code.google.com/p/chromium/codesearch#chromium/src/base/location.h 24 class Location {25 class WEBRTC_EXPORT Location { 25 26 public: 26 27 // Constructor should be called with a long-lived char*, such as __FILE__. -
trunk/Source/ThirdParty/libwebrtc/Source/webrtc/base/messagehandler.h
r212333 r212401 16 16 17 17 #include "webrtc/base/constructormagic.h" 18 #include "webrtc/base/export.h" 18 19 19 20 namespace rtc { … … 23 24 // Messages get dispatched to a MessageHandler 24 25 25 class MessageHandler {26 class WEBRTC_EXPORT MessageHandler { 26 27 public: 27 28 virtual ~MessageHandler(); -
trunk/Source/ThirdParty/libwebrtc/Source/webrtc/base/network.h
r212333 r212401 19 19 20 20 #include "webrtc/base/basictypes.h" 21 #include "webrtc/base/export.h" 21 22 #include "webrtc/base/ipaddress.h" 22 23 #include "webrtc/base/networkmonitor.h" … … 136 137 137 138 // Base class for NetworkManager implementations. 138 class NetworkManagerBase : public NetworkManager {139 class WEBRTC_EXPORT NetworkManagerBase : public NetworkManager { 139 140 public: 140 141 NetworkManagerBase(); … … 201 202 // Basic implementation of the NetworkManager interface that gets list 202 203 // of networks using OS APIs. 203 class BasicNetworkManager : public NetworkManagerBase,204 class WEBRTC_EXPORT BasicNetworkManager : public NetworkManagerBase, 204 205 public MessageHandler, 205 206 public sigslot::has_slots<> { … … 277 278 278 279 // Represents a Unix-type network interface, with a name and single address. 279 class Network {280 class WEBRTC_EXPORT Network { 280 281 public: 281 282 Network(const std::string& name, -
trunk/Source/ThirdParty/libwebrtc/Source/webrtc/base/proxyinfo.h
r212333 r212401 13 13 14 14 #include <string> 15 #include "webrtc/base/cryptstring.h" 16 #include "webrtc/base/export.h" 15 17 #include "webrtc/base/socketaddress.h" 16 #include "webrtc/base/cryptstring.h"17 18 18 19 namespace rtc { … … 26 27 const char * ProxyToString(ProxyType proxy); 27 28 28 struct ProxyInfo {29 struct WEBRTC_EXPORT ProxyInfo { 29 30 ProxyType type; 30 31 SocketAddress address; -
trunk/Source/ThirdParty/libwebrtc/Source/webrtc/base/refcountedobject.h
r212333 r212401 14 14 15 15 #include "webrtc/base/atomicops.h" 16 #include "webrtc/base/export.h" 16 17 17 18 namespace rtc { 18 19 19 20 template <class T> 20 class RefCountedObject : public T {21 class WEBRTC_EXPORT RefCountedObject : public T { 21 22 public: 22 23 RefCountedObject() {} -
trunk/Source/ThirdParty/libwebrtc/Source/webrtc/base/socketaddress.h
r212333 r212401 26 26 27 27 // Records an IP address and port. 28 class SocketAddress {28 class WEBRTC_EXPORT SocketAddress { 29 29 public: 30 30 // Creates a nil address. -
trunk/Source/ThirdParty/libwebrtc/Source/webrtc/base/thread.h
r212333 r212401 94 94 // WARNING! SUBCLASSES MUST CALL Stop() IN THEIR DESTRUCTORS! See ~Thread(). 95 95 96 class LOCKABLE Thread : public MessageQueue {96 class LOCKABLE WEBRTC_EXPORT Thread : public MessageQueue { 97 97 public: 98 98 // Create a new Thread and optionally assign it to the passed SocketServer. -
trunk/Source/ThirdParty/libwebrtc/Source/webrtc/common_types.h
r212333 r212401 28 28 #endif 29 29 30 #ifdef WIN32 30 31 #if defined(WEBRTC_EXPORT) 31 32 #define WEBRTC_DLLEXPORT _declspec(dllexport) 32 33 #elif defined(WEBRTC_DLL) 33 34 #define WEBRTC_DLLEXPORT _declspec(dllimport) 35 #endif 34 36 #else 35 37 #define WEBRTC_DLLEXPORT -
trunk/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/include/video_frame_buffer.h
r212333 r212401 17 17 18 18 #include "webrtc/base/callback.h" 19 #include "webrtc/base/export.h" 19 20 #include "webrtc/base/refcount.h" 20 21 #include "webrtc/base/scoped_ref_ptr.h" … … 57 58 58 59 // Plain I420 buffer in standard memory. 59 class I420Buffer : public VideoFrameBuffer {60 class WEBRTC_EXPORT I420Buffer : public VideoFrameBuffer { 60 61 public: 61 62 I420Buffer(int width, int height); -
trunk/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/libyuv/include/webrtc_libyuv.h
r212333 r212401 19 19 #include <vector> 20 20 21 #include "webrtc/base/export.h" 21 22 #include "webrtc/common_types.h" // RawVideoTypes. 22 23 #include "webrtc/common_video/rotation.h" … … 101 102 // only exception at the time of this writing is 102 103 // VideoCaptureImpl::IncomingFrame, which still needs libyuv::ConvertToI420. 103 int ConvertToI420(VideoType src_video_type,104 WEBRTC_EXPORT int ConvertToI420(VideoType src_video_type, 104 105 const uint8_t* src_frame, 105 106 int crop_x, -
trunk/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/basicpacketsocketfactory.h
r212333 r212401 12 12 #define WEBRTC_P2P_BASE_BASICPACKETSOCKETFACTORY_H_ 13 13 14 #include "webrtc/base/export.h" 14 15 #include "webrtc/p2p/base/packetsocketfactory.h" 15 16 … … 20 21 class Thread; 21 22 22 class BasicPacketSocketFactory : public PacketSocketFactory {23 class WEBRTC_EXPORT BasicPacketSocketFactory : public PacketSocketFactory { 23 24 public: 24 25 BasicPacketSocketFactory(); -
trunk/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/client/basicportallocator.h
r212333 r212401 17 17 18 18 #include "webrtc/p2p/base/portallocator.h" 19 #include "webrtc/base/export.h" 19 20 #include "webrtc/base/messagequeue.h" 20 21 #include "webrtc/base/network.h" … … 23 24 namespace cricket { 24 25 25 class BasicPortAllocator : public PortAllocator {26 class WEBRTC_EXPORT BasicPortAllocator : public PortAllocator { 26 27 public: 27 28 BasicPortAllocator(rtc::NetworkManager* network_manager, -
trunk/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/RTCLogging.mm
r212333 r212401 39 39 NSString* RTCFileName(const char* file_path) { 40 40 NSString* ns_file_path = 41 [[ NSString alloc] initWithBytesNoCopy:const_cast<char*>(file_path)41 [[[NSString alloc] initWithBytesNoCopy:const_cast<char*>(file_path) 42 42 length:strlen(file_path) 43 43 encoding:NSUTF8StringEncoding 44 freeWhenDone:NO] ;44 freeWhenDone:NO] autorelease]; 45 45 return ns_file_path.lastPathComponent; 46 46 } -
trunk/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/videotoolboxvideocodecfactory.h
r212333 r212401 12 12 #define WEBRTC_SDK_OBJC_FRAMEWORK_CLASSES_VIDEOTOOLBOXVIDEOCODECFACTORY_H_ 13 13 14 #include "webrtc/base/export.h" 14 15 #include "webrtc/media/engine/webrtcvideoencoderfactory.h" 15 16 #include "webrtc/media/engine/webrtcvideodecoderfactory.h" … … 17 18 namespace webrtc { 18 19 19 class VideoToolboxVideoEncoderFactory20 class WEBRTC_EXPORT VideoToolboxVideoEncoderFactory 20 21 : public cricket::WebRtcVideoEncoderFactory { 21 22 public: … … 32 33 }; 33 34 34 class VideoToolboxVideoDecoderFactory35 class WEBRTC_EXPORT VideoToolboxVideoDecoderFactory 35 36 : public cricket::WebRtcVideoDecoderFactory { 36 37 public: -
trunk/Source/ThirdParty/libwebrtc/Source/webrtc/video_frame.h
r212333 r212401 12 12 #define WEBRTC_VIDEO_FRAME_H_ 13 13 14 #include "webrtc/base/export.h" 14 15 #include "webrtc/base/scoped_ref_ptr.h" 15 16 #include "webrtc/base/timeutils.h" … … 21 22 namespace webrtc { 22 23 23 class VideoFrame {24 class WEBRTC_EXPORT VideoFrame { 24 25 public: 25 26 // TODO(nisse): Deprecated. Using the default constructor violates the -
trunk/Source/ThirdParty/libwebrtc/libwebrtc.xcodeproj/project.pbxproj
r212397 r212401 42 42 414D71171E4AEAFB0023E526 /* sctp_indata.c in Sources */ = {isa = PBXBuildFile; fileRef = 414D71151E4AEACD0023E526 /* sctp_indata.c */; }; 43 43 414D71191E4AEB110023E526 /* sctp_userspace.c in Sources */ = {isa = PBXBuildFile; fileRef = 414D71181E4AEB110023E526 /* sctp_userspace.c */; }; 44 5C0073031E5513E70042215A /* libboringssl.a in Frameworks */ = {isa = PBXBuildFile; fileRef = 5C63FC601E418411002CA531 /* libboringssl.a */; }; 45 5C0073041E5513E70042215A /* libopus.a in Frameworks */ = {isa = PBXBuildFile; fileRef = 5C4B4A8E1E42C336002651C8 /* libopus.a */; }; 46 5C0073051E5513E70042215A /* libsrtp.a in Frameworks */ = {isa = PBXBuildFile; fileRef = 5C0884D11E4A97E300403995 /* libsrtp.a */; }; 47 5C0073061E5513E70042215A /* libusrsctp.a in Frameworks */ = {isa = PBXBuildFile; fileRef = 5C11A12D1E4577E1004F0987 /* libusrsctp.a */; }; 48 5C0073071E5513E70042215A /* libwebrtcpcrtc.a in Frameworks */ = {isa = PBXBuildFile; fileRef = 5C088C0B1E4AA3B100403995 /* libwebrtcpcrtc.a */; 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